Adaptive differential pulse code modulation ("ADPCM") is well known and widely used for speech transmission over digital links with limited bandwidth because of its bandwidth efficiency. In particular, digital cordless telephone standards such as the CT-2 and DECT are based on 32 Kbps ADPCM speech coding.
Signal level tracking of the ADPCM signal is accomplished through the use of an adaptive quantizer that performs an automatic gain control function before quantization. At each sample of an input analog signal (for example, an RF signal), the adaptive quantizer updates the quantization step size in a recursive manner based on the previous output codeword, effectively adjusting the output level of the encoder. For example, in 4-bit ADPCM, the occurrence of output codeword +7 or -7 increases the step size for the next input sample substantially while the occurrence of codeword 0 reduces it moderately.
FIG. 1 shows a typical distribution of ADPCM codewords. With transmission bit errors, the codeword distribution at the receiver tends to flatten as indicated. That is, the bit errors tend to, on average, increase the magnitude of the codeword input at the receiver. As a result, there are more codewords that increase the quantizer step size at the receiver than there are at the transmitter, resulting in an overall output level increase.
ADPCM speech encoding provides many advantages for speech transmission. However, there are also deficiencies with such a coding scheme. For example, if fading of the digital radio links occurs, transmitted bits experience occasional high bit error rates as a result.
When such transmission bit errors occur, an ADPCM decoder not only introduces noise to the output, it also amplifies that noisy output. At bit error rates greater than five percent, the ADPCM output noise level is too high for normal listening. In order to overcome this problem, muting is normally applied to the ADPCM output whenever the RF signal strength falls below a certain threshold level, cutting off the ADPCM output entirely. The threshold level is chosen based on the likelihood of a bit error rate at the low level and the associated noise level at the output.
This approach, however, has several deficiencies. One of the shortcomings is the sudden signal dropout that occurs due to muting; a listener on the receive end suddenly gets no input. Another shortcoming is the lack of indication to the listener as to the link condition; when signal dropout is experienced, it is not known by the listener whether this is due to RF fading or due to some other problem, such as a hardware malfunction. A third shortcoming is the coarse nature of judging the link condition based solely on the signal strength. This method of noise reduction is actuated based solely on the input analog signal strength, when other characteristics should also be taken into consideration in evaluating the quality of the link. As the normal solution is currently implemented, it is possible to enable muting even if the link condition is acceptable. Further, such a solution does not compensate at the receive side if there is a high bit error rate and the input analog signal is of adequate strength, that is, if the bit error rate is unrelated to signal strength.
Several attempts have been made to rectify shortcomings in ADPCM speech encoding. For example, U.S. Pat. No. 4,354,273 to Araseki et al. discloses an ADPCM system for speech transmission. The coefficient of the synthesis filter in both the transmitter and the receiver of the Araseki system is varied in accordance with the normalized error rather than with the error itself, providing greater frequency band compression and preventing transmission errors from rendering the synthesis filter unstable.
U.S. Pat. No. 4,571,736 to Agrawal et al. discloses an ADPCM system which overcomes feedback interference introduced in the code by periodically not transmitting a percentage of speech data samples and instead replacing the samples by their estimates. The coding process then continues in a normal fashion. The estimate is established on the basis of autocorrelation statistics of the speech data samples. At the receiving end of the communication system, the replaced sample is estimated using the same process and is further estimated using delayed interpolation. This technique is useful in achieving gradual degradation on the data channel, but is most useful when the bit rate is 24 Kbps or less.
U.S. Pat. No. 4,088,876 to Rege discloses a recirculating memory circuit made of shift registers that may be accessed in parallel. The circuitry detects the occurrence of a series of ones or a series of zeros and relates such an occurrence to a high probability of an error in that shift register. The corresponding bits in the data segment being accessed can then be corrected by complementing the appropriate error bits.
U.S. Pat. No. 4,481,629 to Hatata et al. discloses an abnormal signal detecting device having a signal sampling and conversion circuit for sampling a number of data points along an analog signal and generating digital sampled data signals representative of the analog signals. The device also stores the sampled signal and compares consecutive samples. An output indicating an abnormal condition is generated whenever at least three consecutive digital samples are the same.
U.S. Pat. No. 4,502,143 to Kato et al. discloses an encoder which tests for consecutive bits having the same value. After a certain predetermined number of bits having the same value have been detected, a bit having the complement value is inserted into the bit stream in order to keep the average signal level constant. The encoder is useful for digital communication at a rate of higher than 100 Mbps.
U.S. Pat. No. 4,747,112 to Blondeau, Jr. et al. discloses a decoding method for T1 line zero bit suppression in which an encoded T1 line frame that has a number of bundled channels is received. Indicator bits of the received T1 line frame are tested to determine whether any channel of the frame has been altered for the transmission. If alteration is determined, the channels in the bundle that have been altered are isolated and the contents of each altered channel are replaced with all zeros and the T1 line frame is transferred for further processing.
U.S. Pat. No. 4,794,604 to Gorshe discloses a system for exploiting the characteristics of the Zero-Byte Time Slot Interchange (ZBTSI) algorithm by using the relationship of the data octet and adjacent octets to detect error conditions such as violations of the DS1 ones density criteria for detection of transmission channel errors in a ZBTSI decoder. This relationship is used as part of an optimized partial error correction technique to minimize error multiplication in the transmission channel.
U.S. Pat. No. 4,853,931 to Gorshe discloses a violating all-zero octet detector for a ZBTSI clear channel data transmission system. The input data are scanned for zero strings of data that could combine with an all-zero octet to violate the zero string criterion of the system.
None of the above references describes a system that effectively solves the problem of increased noise levels on a digital channel in a fading RF environment utilizing ADPCM encoding. Such a system should gradually degrade the ADPCM output as bit error rates increase, effectively keeping the received noise level in check. Such degradation of the signal should be based at least in part on an estimate of the bit error rate, rather than solely on the input RF signal level.